In general, conditions such as sufficient bandwidth, a small delay and a small packet loss are needed to successfully transmit multimedia data through a wired Internet protocol (IP) network. Unfortunately, the current network layer in the related art wired IP networks cannot provide a function suitable to meet quality of service (QoS) required for video transmission. Therefore, the QoS should be secured by a higher layer of the network layer. To this end, a real-time transport protocol (RTP) and a real-time transport control protocol (RTCP) on a transport layer have been proposed.
The RTP is an Internet protocol for real-time multimedia data such as real-time audio and video. Although the RTP does not ensure real-time transmission of data, using the RTP, application programs for transmission/reception in a real-time multimedia data communication system can support streaming data. The RTP is commonly executed over user datagram protocol (UDP).
The RTCP is a protocol used to maintain the QoS of the RTP. While the RTP is related to data transmission, the RTCP relates to monitoring of the data transmission and the transmitting of session-related information. RTP nodes send RTCP packets to one another, in order to analyze a network state and to periodically report whether the network is congested.
By the use of the RTP and the RTCP, network characteristics related to time limits in transmitting multimedia data may be considered. Accordingly, packet losses generated in the network may be remedied.
In a general multimedia data communication system, a multimedia application program (application layer) detects the network state through the RTCP and controls an encoding rate of real-time multimedia data to be transmitted. The controlling of the encoding rate of the real-time multimedia data is made through transmission rate control.
A general method of estimating an effective transmission rate by using network state information by the RTCP is provided in Equation 1.
                              R          ⁡                      (            t            )                          =                              1.22            ×            s                                              RTT              ⁡                              (                t                )                                      ×                                          p                ⁡                                  (                  t                  )                                                                                        (                  Equation          ⁢                                          ⁢          1                )            
The R(t) indicates an effective transmission rate, the p(t) indicates a packet loss rate and is obtained by the RTCP transmitted from a receiving side. The RTT(t) indicates a round-trip delay time, and the ‘s’ indicates size of a packet.
When the round-trip delay time (RTT(t)) and the packet loss rate (P(t)) are given, an effective transmission rate is estimated by Equation 1. When the packet size ‘s’ is fixed, the estimated transmission rate varies according to the RTT(t) and P(t).
FIG. 1 shows a change in an available transmission rate estimated by Equation 1, when the P(t) and the packet size ‘s’ are fixed and the RTT(t) is linearly changed. When the packet loss rate is fixed such that p(t)≡0.015 and s≡625 and the RTT(t) is linearly increased from 80 ms to 380 ms, the minimum transmission rate is 50 kbps and the maximum transmission rate is 500 kbps.
As shown in FIG. 1, as the RTT(t) increases, the estimated available transmission rate decreases. FIG. 2 shows a change in an available transmission rate estimated by Equation 1, when the RTT(t) and the packet size ‘s’ are fixed and the P(t) is linearly changed. When the RTT(t) is fixed to 100 ms, the packet size ‘s’ is fixed to 625 and the P(t) is changed from 0.1% to 20%, the minimum transmission rate of the user is 50 kbps and the maximum transmission rate is 500 kbps.
As shown in FIG. 2, in the related art method for estimating an effective transmission rate according to Equation 1, a time out occurs when large packet losses are not accounted for. That is, when the packet loss rate is small, the available transmission rate is estimated at an appropriate value according to the related art. However, if the packet loss rate is large, the available transmission rate is overestimated.
For example, when the packet loss rate is 10% (considerably large) the available transmission rate is undesirably overestimated at about 200 kbps by the related art estimation method of the transmission rate. Thus, the general estimation method of the related art rate is disadvantageous in that the available transmission rate is overestimated when the packet loss rate is large and the network congestion cannot be quickly resolved due to the above inaccuracies.
A solution is needed to overcome the above stated problem.